Apparatus and method for improved signal fade out for switched audio coding systems during error concealment
US-9916833-B2 · Mar 13, 2018 · US
US10763885B2 · US · B2
| Field | Value |
|---|---|
| Publication number | US-10763885-B2 |
| Application number | US-201916675051-A |
| Country | US |
| Kind code | B2 |
| Filing date | Nov 5, 2019 |
| Priority date | Nov 6, 2018 |
| Publication date | Sep 1, 2020 |
| Grant date | Sep 1, 2020 |
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In an embodiment, a method includes: receiving an audio frame; decomposing the received audio frame into M sub-band pulse-code modulation (PCM) audio frames, where M is a positive integer number; predicting a PCM sample of one sub-band PCM audio frame of the M sub-band PCM audio frames; comparing the predicted PCM sample with a corresponding received PCM sample to generate a prediction error sample; comparing an instantaneous absolute value of the prediction error sample with a threshold; and replacing the corresponding received PCM sample with a value based on the predicted PCM sample when the instantaneous absolute value of the prediction error sample is greater than the threshold.
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What is claimed is: 1. A method comprising: receiving an audio frame; decomposing the received audio frame into M sub-band pulse-code modulation (PCM) audio frames, wherein M is a positive integer number; predicting a PCM sample of one sub-band PCM audio frame of the M sub-band PCM audio frames; comparing the predicted PCM sample with a corresponding received PCM sample to generate a prediction error sample; comparing an instantaneous absolute value of the prediction error sample with a first threshold; and replacing the corresponding received PCM sample with a value based on the predicted PCM sample when the instantaneous absolute value of the prediction error sample is greater than the first threshold. 2. The method of claim 1 , wherein M is 32. 3. The method of claim 1 , further comprising: decomposing a PCM audio frame into the M sub-band PCM audio frames; encoding the M sub-band PCM audio frames into an encoded audio frame; transmitting the encoded audio frame; and unpacking the encoded audio frame to generate the audio frame. 4. The method of claim 1 , further comprising: computing an RMS value of the one sub-band PCM audio frame; and determining the first threshold based on the computed RMS value. 5. The method of claim 4 , further comprising: computing respective RMS values of each of the M sub-band PCM audio frames, wherein computing the respective RMS values of each of the M sub-band PCM audio frames comprises computing the RMS value of the one sub-band PCM audio frame; determining respective thresholds based on the computed respective RMS values of each of the M sub-band PCM audio frames, wherein determining the respective thresholds comprises determining the first threshold; comparing respective instantaneous absolute values of respective prediction error samples of each of the M sub-band PCM audio frames with the respective thresholds, wherein comparing the respective instantaneous absolute values of the respective prediction error samples of each of the M sub-band PCM audio frames with the respective thresholds comprises comparing the instantaneous absolute value of the prediction error sample of the one sub-band with the first threshold; and replacing corresponding received PCM samples of each of the M sub-bands with respective values based on predicted PCM samples when the respective instantaneous absolute values of the respective prediction error samples are greater than the respective thresholds, wherein replacing the corresponding received PCM samples of each of the M sub-bands with the respective values based on the predicted PCM samples comprises replacing the corresponding received PCM sample of the one sub-band with the value based on the predicted PCM sample. 6. The method of claim 4 , wherein the percentage is between 70% and 200%. 7. The method of claim 1 , further comprising: generating a prediction error frame comprising prediction error samples corresponding to received PCM samples in the one sub-band PCM audio frame; computing an RMS value of the prediction error frame; and determining the first threshold as a percentage of the computed RMS value. 8. The method of claim 7 , wherein the percentage is between 70% and 200%. 9. The method of claim 1 , further comprising: receiving an encoded frame using an MPEG Version 1 Layer II (MP2) decoder; and generating the audio frame with the MP2 decoder. 10. The method of claim 1 , wherein predicting the PCM sample comprises using Levinson's method. 11. The method of claim 1 , further comprising: performing spectral analysis of the one sub-band PCM audio frame; determining a predicting order based on the performed spectral analysis; and predicting the PCM sample using a linear predictor of the determined predicting order. 12. The method of claim 1 , wherein the one sub-band PCM audio frame comprises a frequency span between 3000 Hz and 3750 Hz. 13. The method of claim 1 , wherein the audio frame is an AM High Definition (AM HD) audio frame, an FM High Definition (FM HD) audio frame, a digital audio broadcasting (DAB) audio frame, a DAB+ audio frame, or a Digital Radio Mondiale (DRM) audio frame. 14. A digital audio broadcasting (DAB) receiver comprising: a frame unpacking block configured to receive an encoded frame; a reconstruction block configured to generate M sub-band frames based on the encoded frame; a linear predictor configured to receive one sub-band frame of the M sub-band frames and to generate a predicted sample based on a corresponding received sample; a first block configured to generate a prediction error based on the predicted sample and the corresponding received sample; a comparator configured to assert a flag when an instantaneous absolute value of the prediction error is higher than a threshold; and a selector block configured to select the predicted sample when the flag is asserted. 15. The DAB receiver of claim 14 , further comprising a second selector block comprising: a selection input configured to receive a frame error flag; a first input configured to receive original PCM samples; and a second input configured to receive PCM samples from the selector block; and an output, wherein the second selector block is configured to: connect the output of the second selector block to the first input of the second selector block when the frame error flag is deasserted, and connect the output of the second selector block to the second input of the second selector block when the frame error flag is asserted. 16. The DAB receiver of claim 15 , wherein the selection input of the second selector block is configured to receive the frame error flag from the reconstruction block. 17. A method of concealing an error in a pulse-code modulation (PCM) audio frame encoded with an MPEG Version 1 Layer II (MP2) encoding scheme, the method comprising: receiving the PCM audio frame; decomposing the received PCM audio frame into 32 sub-band PCM audio frames; predicting a PCM sample of one sub-band PCM audio frame of the 32 sub-band PCM audio frames using a linear predictor based on PCM samples of the received PCM audio frame; comparing the predicted PCM sample with a corresponding received PCM sample to generate a prediction error sample; comparing an instantaneous absolute value of the prediction error sample with a threshold; and replacing the corresponding received PCM sample with the predicted PCM sample when the instantaneous absolute value of the prediction error sample is greater than the threshold. 18. An audio decoder comprising: a frame unpacking block configured to receive an encoded frame; a reconstruction block configured to generate M sub-band frames based on the encoded frame; and an error detection and concealment block comprising: a linear predictor configured to receive one sub-band frame based on the M sub-band frames and to generate a predicted sample based on a corresponding received sample, an adder block configured to generate a prediction error based on the predicted sample and the corresponding received sample, a comparator configured to assert a flag when an instantaneous absolute value of the prediction error is higher than a threshold, and a selector block configured to select the predicted sample when the flag is asserted. 19. The audio decoder of claim 18 , wherein the audio decoder is of the MPEG Version 1 Layer II (MP2) type. 20. The audio decoder of claim 18 , further comprising: an MPEG Version 1 Layer II (MP2) decoder that comprises the
with adaptable step size, e.g. adaptive differential pulse code modulation [ADPCM] · CPC title
Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing · CPC title
Conversion to or from differential modulation with several bits only, i.e. the difference between successive samples being coded by more than one bit, e.g. differential pulse code modulation [DPCM] (H03M7/3004 takes precedence; voice coding G10L19/00; image coding H04N19/00) · CPC title
with adaptive feedback · CPC title
the extracted parameters being prediction coefficients · CPC title
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